1. Field of Invention
The present invention relates generally to the field of telephone services. More specifically, the present invention relates to accessing mobile network voice services from an IP client.
2. Discussion of Prior Art
A mobile operator's network also known as a Public Land Mobile Network (PLMN) typically comprises a universal terrestrial radio access network (UTRAN) and a circuit switched wire-line core network as illustrated in FIG. 1. The UTRAN carries the radio signals from the cell phones to the core network and back. The core network determines the location of mobile users and performs the required switching and service delivery functions using circuit switched or packet based core switches to route calls. The Third Generation (also known as 3G) networks describe a packet switched core network as an alternative to traditional Second Generation (2G) circuit switched core networks.
Referring to FIG. 1, the universal radio access network, UTRAN, 506 carries the radio frequency (RF) signals from the wireless cell phone 102 to the switches that form the mobile operator's core network 305. A UTRAN may use Time Division Multiple Access (TDMA) or Code Division Multiple Access (CDMA) for handling RF signals. A UTRAN base station system (BSS) 808 communicates with cell phone 102 using allocated radio frequencies, and using Base Station System Application Part (BSSAP) protocol or other similar protocols, it sends location information about the mobile user 102 to the Mobile Switching Center (MSC) 505, which is a key component of the core network 305. The core network 305 comprises many MSCs and facilities that interconnect them.
There is a functionality embedded within the MSC known as Visitor Location Register (VLR), which retrieves information about the location of mobile user 102, stores it locally, and updates the centralized register known as Home Location Register (HLR) 501 using Mobile Application Part (MAP) protocol or other similar protocols. Doing so, the centralized HLR has up to date knowledge about which VLR/MSC each mobile user is currently attached to while they roam. Protocols such as MAP are specifically defined for GSM networks and considered to be part of SS7 signaling, but there are equivalent protocols in other types of networks, all defined by proper standards bodies (ETSI, ANSI etc.).
If mobile user 103 initiates a phone call to mobile user 102, it first reaches VLR/MSC 505 or the MSC to which mobile user 103's BSS connects to (in this scenario it is the same VLR/MSC as the one user 102 attaches to) so as to obtain location information about user 102. If the location of user 102 is in the local database of the VLR, then that MSC can process the call. Otherwise, it initiates a Mobile Station Roaming Number (MSRN) request using MAP protocol to HLR 501 to learn which MSC the user 102 is attached to. Such a request also allows the HLR to send other information about subscriber's services to the MSC. The HLR stores subscriber service information as well as subscriber location. The subscriber service information is provisioned into the HLR using a provisioning system. If the HLR sends information about the services associated with the caller 103 or called 102 (such as prepaid billing, or number translation), the VLR/MSC sends an Intelligent Network Application Protocol (INAP) request to Service Control Point (SCP) 802 for instructions to handle a call that has associated intelligent or supplementary services. In prior art, the SCP 802 is where subscriber's services are processed. In response to the INAP message, SCP 802 may contact local databases to perform appropriate address translations, or may contact the billing system for a prepaid charging request. For services such as prepaid, SCP 802 maintains the call state during the call in order to deduct appropriate amounts from user 103's prepaid billing account until the call terminates.
The VLR/MSC 505 uses ISDN User Part (ISUP) signaling protocol towards the other switches in the network for call path establishment. The HLR may be provisioned manually or automatically with subscription based services. The SCP may also be configured manually or electronically for processing the calls for service delivery. All these components and service delivery steps are prior art and well understood.
During the last decade, the Internet Engineering Task Force (IETF) has developed protocols to carry voice along with data on IP networks. Recently, millions of people have started using the Internet for voice in addition to data. Although phone calls may originate on a terminal attached to the Internet, because the called party will most likely be attached to a non-IP (legacy) network (mobile or fixed networks), a translation gateway is needed to bridge Internet telephony to legacy telephony both for signaling and bearer translations. This translation gateway is known in prior art as a softswitch.
In the softswitch approach, the calling party subscribes to services of a Voice over IP (VoIP) operator (such as Net2Phone® or Vonage®), who has a private Internet backbone, which is also attached to the public Internet to allow access from the public Internet, and has interconnectivity to other operators network to have access to users on other operator's circuit switched network. The softswitch is most commonly owned by the VoIP operator and has interface with mobile or fixed networks.
In order to contrast the above operations with that of the fixed environment, a VoIP operator's (such as Vonage® or Net2Phone®) network is shown in FIG. 2. The Session Initiation Protocol (SIP) as described in RFC 3261 may be used as a VoIP signaling protocol. In FIG. 2, the VoIP terminals 101 that can make phone calls are referred as “SIP clients”. It does not preclude, however, that the client may use another IP telephony protocol.
In prior art, the VoIP Operator's network comprises a fixed access network 302a, a fixed core network 302b and at least one softswitch, which is attached to both the VoIP operator's IP network, and simultaneously, to the circuit switched Public Switched Telephony Network (PSTN). The access network may use dial, fixed wireless, cable, ADSL, private line or other narrowband or broadband technologies depending on VoIP operator's choice. In some cases, the VoIP operator may not have an access network. Meaning they rely on the Internet Service Provider's access network to offer the service. For example, if a cable operator such as Comcast® is the VoIP operator, they use the cable network they own to deliver cable TV service for VoIP, and they use an IP backbone network. Another example VoIP operator is Vonage® who does not own an access network, and simply uses the internet access that an ADSL or Cable Operator such as Comcast® delivers to home. In this scenario, Vonage® provides an access box from Motorola that provides a connection to the internet at home and the telephone on another termination to make phone calls.
The VoIP operator's attachment to the PSTN can be performed via connecting to one of the telephone service provider. Connecting to one provider, in general, provides access to many other operators' network through that provider's interconnectivity agreements with other operators and physical connectivity. The softswitch components are prior art and hence will not be discussed in detail in this application. When SIP client 101 attaches to VoIP operator 302's network and calls a phone client attached to the PSTN 601, the SIP signaling originated in SIP Client 101 is terminated on SIP Proxy server 701 in VoIP operator's network or it may alternatively be embedded with softswitch.
Softswitch performs the appropriate signaling translations between IP to SS7/ISUP telephony signaling, translations between IP voice bearers to circuit switched voice bearers, and other well-known functions such as control of the Media Gateway subcomponent of the softswitch using Media Gateway Control Protocol (MGCP) or other protocols. All these components and protocols are well defined in prior art.
If the called party in this scenario is a mobile terminal, then the routing to the appropriate VLR/MSC is performed simply by the interconnection between the PTSN telephony operator and the PLMN. If the called or calling party has subscription-based services, then SCP 802 provides the needed functionality using either the SIP protocol or the INAP protocol between the Softswitch and the SCP.
From the service delivery perspective, the SIP client is a subscriber of the VoIP operator. Hence, the client has access to only the services that the VoIP operator provides. The SIP client's telephone number is provided by the VoIP operator who has a numbering pool.
The SIP client is attached to a fixed network operator. In the case of a subscriber of a VoIP operator initiating a voice call from his/her PC 101 attached to the public Internet 301 at a hotel lobby or Internet cafe, the SIP client of the PC connects to the VoIP operator's network 302. The SIP protocol requires a “SIP proxy server” 701 to intercept and process the SIP calls for signaling. The SIP client on PC 101 has to find the SIP proxy in the VoIP operator's network 302 to process the call. The SIP client finds the IP address of such a SIP proxy by querying the Domain Name Services (DNS) 702 using its SIP domain name as the handle. The SIP client performs the DNS lookup to receive the IP address corresponding to the SIP domain name of the SIP proxy server 701. Having the IP address, the SIP client can now connect to the SIP proxy server 701 in the VoIP operator's network 302 which can further process the phone call. The SIP proxy routes the call to a signaling gateway 307 which can communicate with the SIP protocol on one hand within the IP network and with the SS7 signaling protocol with the non-IP networks on the other. These are links 405 (SIP) and 407 (SS7) respectively. Doing so, the signaling path can be extended to the circuit switched network to find the called party which is on that network. Once the signaling is completed, the voice calls get routed from the calling party to the called party by first traveling the IP network if the form of RTP packets and then through a media gateway 309 which translates the RTP packets to circuit switched voice traffic and finally through the mobile operator's circuit switched network to reach the called party attached to it. The media gateway 309 is controlled by media gateway controller 306 using protocols such as MGCP (multimedia gateway control protocol) and MEGACO (media gateway control). The totality of components 306, 307, and 309 along with SIP Proxy Server 701 is sometimes referred in prior art a “softswitch”.
Unfortunately, none of these solutions have an ability to offer seamless services to the users. Today, users cannot utilize their services in mobile networks from the Internet, and even more importantly they have to carry the burden of maintaining different telephone numbers, accounts and services on different networks. Simply, softswitch brought convergence in the networking layer, but not in the services layer.
Whatever the precise merits, features, and advantages of the above discussed systems, none of them achieves or fulfills the purposes of the present invention.